search for: conference

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has been changed. I m not providing the sources now, probably tom...

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at confere...

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, all...

I have a site (for an academic conference) that I want to convert to RoR - both as a learning exercise and to ease further updates and development. There are a number of child pages for each conference (past and future), but these pages are different for each conference (not same child pages for each conference). The URLs are along the...

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far...

...ple into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf looks like: [general] autofallthrough=yes static=no writeprotect=no extenpatternmatchnew=yes clearglobalvars=no [conference-calls] exten => s,1,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Background(thank-you-for-calling) exten => s,n,Background(conference-reservations) exten => s,n,Wait(2) exten => s,n,Background(enter-conf-pin-number) exten => s,n,WaitExt...

...ple into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf looks like: [general] autofallthrough=yes static=no writeprotect=no extenpatternmatchnew=yes clearglobalvars=no [conference-calls] exten => s,1,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Background(thank-you-for-calling) exten => s,n,Background(conference-reservations) exten => s,n,Wait(2) exten => s,n,Background(enter-conf-pin-number) exten => s,n,WaitExt...

...2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines file includes definitions for the prefered channel and context *************************************************** 5. Call history report. Support for this feature requires the php script ./lib/cbEnd.php be running at all times. This also requires a new table in the meet...

How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(<terminal>) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan d...

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating my EAGI script, etc) the log fills up with error messages and the channel disconnects immediately. What could be wrong? My configuration is mostly duplicated from the old 1.4 config - has somethi...

Hi there, I'm using asterisk meetme function like: exten => 9070,n,MeetMe(|dcM) and everything works pretty well. But I would like to add a review of the entered conference number before the user jumps into the conference. Somthing like: *:"Please enter the conference number followed by the hash key" (works) U: 123456# (works) *: "You are entering conference number 123456" (missing) *: "You are currently the only person in this conference&quo...

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof...

...te a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? * Asterisk on SlashDot * Asterisk on the O'Reilly Open Source Convention * Writing community documentation - asteriskdocs.org * Application of the week: The MeetMe conference system * Alternatives to the standard MeetMe *** Looking beyond Asterisk 1.0/1.1 - what's up? ------------------------------------------------- At Astricon in September we'll arrange a developer's meeting. This is the first time we gather all Asterisk programmers and documentation w...

So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1 port FXO card). So,...

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set a PIN. Hangup. Then user...

...ere's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All callers automatically are joined into the same conference and are able to speak/hear each other at the same time they're listening to the stream. The tricky part: A caller may now press some digits on their their keypad, like 1, or 2, and this will have the effect that the volume of the stream is increasing or decreasing. BUT: When they adjust...

...he server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP devices are able to call into a MeetMe conference without issue. However, when I attempt to transfer an inbound call from one of my SIP devices to a MeetMe conference, the call is dropped. If I complete the transfer while the "You are currently the only person in this conference" prompt is playing, the call will successfully make it into...

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconn...

Hello, I was wondering if it is possible within Asterisk to be in many meetme conferences at the same time. This would be sort of broadcasting over all the conferences at once. Thanks, Denis

I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the people that nor...